The vast majority of vehicles currently in use incorporate vehicle communication systems for receiving signals. For example, vehicle audio systems provide information and entertainment to many motorists daily. These audio systems typically include an AM/FM radio receiver that receives radio frequency (RF) signals. These RF signals are then processed and rendered as audio output. Originally live multimedia content, such as real time audio/video, was delivered to In-Dash Multimedia Players by analog wireless modulation like FM radio or analog TV. These analog broadcast networks allow the use of only a limited number of channels, are subject to cross channel interference and have a poor signal quality. Satellite or ground based digital broadcast networks such as Digital Radio and Digital TV in various embodiments such as DAB, HD Radio, XM Radio, Sirius, Sat TV, DVB, DTT and DMB have removed or reduced these limitations using digital transmission in various forms and with various methods to deliver live streaming media (LSM) such as audio/video, but these systems do not have a provision to assure the continuity of reception when the vehicles moves into areas of signal obstruction or lack of coverage. As such areas can be quite large, or the vehicle can be in such areas for a long time (e.g. because of slow moving traffic in long tunnels, urban canyons, mountainous terrain), interruptions of the LSM can be quite long and annoying. Recently, digital wireless networks (DWLN) capable of high data rate transfers such as those found in later generation mobile telephony (e.g. GPRS, EDGE, CDMA, UMTS, WCDMA, LTE) or in Internet oriented cellular networks such as WiFi and WiMax have been widely deployed and are continuously expanding in coverage and data bandwidth. The DWLN can also be used to stream media such audio/video to a vehicle with protocols such as radio or video on Internet, but the DWLN are subjected to similar limitations of signal obstruction, or lack of coverage, as the other systems. The DWLN, differently from the digital and analog broadcast networks, provide a bidirectional communication with the users thus allowing the retransmission of missing data and also the introduction of functions that can relate the LSM to the specific user. For purposes of clarity within this document we refer to LSM, or live streaming media, as material that originates from a source having a real-time nature, such as a radio or TV broadcast. The LSM has as its source a system or arrangement that by definition can only be transmitted to users as fast as the material is generated; for example, a disk jockey speaking into a microphone. The two fundamental ways of transmitting LSM in digital form to the users are by: (i) the digital-broadcast networks such as used in Digital Radio and Digital TV where the same stream of data is transmitted in real time, generally with some degree of error correction, to all users that are able to tune-in and decode the signal, and (ii) the DWLN that can deliver media streams such as internet radio or internet TV to users that are connected to the network. One difference between the two types of transmission is that in the digital-broadcast networks the number of users is unlimited while in the DWLN the maximum number of users that can receive the LSM depends on the available data bandwidth, the requested quality of service, and the system and method used for the transmission. A live streaming multimedia service to in-car users via the DWLN is the focus of this patent application. The DWLN are generally composed of a wireless segment such as an Access Network, and a land segment such as a Core Network, to connect the users to the source of the real time media. The LSM are generally transmitted over the DWLN using one or more protocols of the Internet Protocol Suite such as TCP or UDP or protocols specific to the DWLN such as described in the 3GPP or IMS specifications.
Today most Internet streamed audio and video are compressed to be listenable or viewable to less than 64,000 bits per second (bps) bandwidth and when the available bandwidth falls below the streaming bit rate the playback is interrupted. The nominal bandwidth of the DWLN is generally higher, such as from 200,000 to 7,000,000 bps, but the actual bandwidth can vary from 0 bps (i.e. the transmission is interrupted) to the nominal maximum bandwidth. The reductions of bandwidth can be of short duration (e.g. less than 100 sec) or long duration (e.g. between 100 sec and 15 minutes). The short duration reductions of bandwidth are caused by many factors such as congestion of the wireless or land segment of the network, interference and transmission quality issues. The long duration reductions of bandwidth are generally caused by problems in the wireless segment of the network such as obstructions, interference, cell changeover and distance from the transmitters. Depending on the used protocol the reduction of bandwidth can cause only a delay (e.g. as in TCP/IP protocol) or also a loss of data (e.g. in UDP/IP protocol). Each of these factors can cause delays and interruptions in the transmission of data thus preventing the user from being able to listen to or view uninterrupted LSM unless some special provisions have been incorporated. The short term interruptions are commonly referred to as ‘dropouts’, meaning that the data flow to the user has been shortly interrupted (i.e., the audio ‘drops out’). Dropouts can be extremely annoying, for example, while listening to music. The long term interruptions are even more annoying and within this document we refer to them as outages. If a reliable protocol is used (e.g. TCP, SCTP) a common solution to the problem of dropout is to use a pre-buffering technique to store up enough audio or video data in the user device so that it can play the audio or video with continuity. When the user connects to the network, audio/video output at the user's system is delayed while the user's buffer is filled to a predetermined level. This process requires the user to wait until enough of the media file is buffered in memory before listening or viewing can begin. Typical pre-buffering wait times range from 10 to 30 seconds, determined by the vendor providing the audio or video media. This pre-buffering process avoids dropouts due to transmission delays shorter than the pre-buffering time, but it is not effective against loss of streaming data. In these systems the audio or video data is delivered from the source at the rate it is to be played out. If, for example, the user is listening to an audio stream encoded to be played-out at 24,000 bits per second, the source sends the audio data at the rate of 24,000 bits per second. Provided that the user waits 10 seconds, and the receipt of the buffering data has not been interrupted, there is enough media data stored in the buffer to play for 10 seconds. Cumulative delays in the receipt of audio/video data longer than 10 seconds cause the buffer to deplete. Because transmission of audio/video media data to the user takes place at the rate it is played out, the user's buffer level can never be increased or replenished while it is playing. This method can be applied only to dropouts and streaming media of limited duration, otherwise the size of the pre-buffer has to be increased and the time required to fill it would require the user to wait for an uncomfortable time. U.S. Pat. No. 7,716,358, to Price discloses a method and a system that exploits the fact that the data bandwidth of Internet is higher than the data bandwidth required by the streaming media. Price's patent uses a double buffer, one buffer at the source (e.g. in a server) and one buffer at the user, to eliminate the dropouts due to delays or loss of data without the need for an initial user wait time. This patent is focused on fixed Internet networks and does not consider issues typical of DWLN such as the bandwidth limitations related to the number of connected users in the mobile cell or the functions that relate to the specific users and their mobility.
Conventional streaming media systems may incorporate buffering systems for programmatic purposes. For example, the system may buffer media data at the server for the purpose of packet assembly or disassembly. Media data may also be buffered at the server to permit programming conveniences such as dealing with chunks of data of a specific size or offer time-shift functions to the user.
The sending of audio or video files via a network is known in the art. U.S. Pat. No. 6,029,194 to Tilt describes a media server for the distribution of audio/video over networks, in which retrieved media frames are transferred to a FIFO buffer. A clock rate for a local clock is adjusted according to the fullness of the buffer. The media frames from the buffer are sent in the form of data packets over the networks in response to interrupts generated by the local clock. In this manner, the timing for the media frames is controlled by the user to assure a continuous stream of video during editing. U.S. Pat. No. 6,014,706 to Cannon, et al. discloses an apparatus and method for displaying streamed digital video data on a client computer. The client computer is configured to receive the streamed digital video data from a server computer via a computer network. The streamed digital video data is transmitted from the server computer to the client computer as a stream of video frames. U.S. Pat. No. 6,002,720, to Yurt, et al. discloses a system of distributing video and/or audio information wherein digital signal processing is employed to achieve high rates of data compression. U.S. Pat. No. 5,923,655, to Veschi et al. discloses a system and method for communicating audio/video data in a packet-based computer network wherein transmission of data packets through the computer network requires variable periods of transmission time. U.S. Pat. No. 5,922,048 to Emura discloses a video server apparatus having a stream control section which determines a keyframe readout interval and a keyframe playback interval that satisfy a playback speed designated by a terminal apparatus. U.S. Pat. No. 6,014,694 to Aharoni, et al. discloses a system and method for adaptively transporting video over networks, including the Internet, wherein the available bandwidth varies with time. U.S. Pat. No. 6,378,035 to Parry et al. discloses a system and method for managing at the user end a buffer for streaming information. U.S. Pat. No. 7,280,662 to Walker et al. discloses a system to use a time shifting buffer to manage the availability of received data in a satellite-based digital audio radio. U.S. Pat. No. 6,034,746 to Desai et al. discloses a method and system to insert additional data such as commercials in a media stream. US Pat. Application No. US 2003/0139966 to Sirota et al. discloses a method for replacing pre-cached advertisement into a media stream. US Pat. Application No. US 2009/0260030 to Karlsson et al. discloses a mechanism to replace default advertisements with other advertisements in a media stream. US Pat. Application No. US 2005/0094815 to Walker et al. discloses various embodiments that manage the availability of a media stream in a time-shift buffer. US Pat. Application No. US 2008/0126420 to Wright et al. discloses systems and methods to meter media content presented on a wireless communication device. There remains a need in the art for a method and system for In-Dash Multimedia Players or Portable Multmedia Player for automotive, aviation, boating, and personal use that, exploiting the services and capabilities of the DWLN, such as 3G networks, and buffering the live stream data on both the both the transmitting and receiving ends, combines in a novel synergistic integration a suite of new and known features such as immediate and uninterrupted listening/viewing of live streaming media by the user with or without optimization of data bandwidth, and the provision to customize the commercial messages according to the user location, and the capability of reporting the customer choices and habits, and the automatic erase of the LSM after the time-shift time to benefit from copyright agreements.